Trying to make Opus Codec work with Yealink+Asterisk 13.12.1

Yealink released a new firmware supporting Opus codec.
I am trying to make a T23G work with FreePBX Distro 10.13.66-17 and Asterisk 13.12.1, which includes support for Opus.
Do I have to do anything more except for the upgrade to 13.12.1? Opus is selectable from FreePBX GUI/advanced SIP settings, but when calling into an internal conference, I get:
[2016-10-31 23:35:16] WARNING[13386][C-0000001a]: codec.c:397 ast_codec_samples_count: Unable to calculate samples for codec opus.

Opus doesn’t show under core show translation, but it is listed under core show codecs.

localhost*CLI> core show codecs
Disclaimer: this command is for informational purposes only.
It does not indicate anything about your configuration.

  30 image png          png              (PNG Image)
   5 audio g726         g726             (G.726 RFC3551)
   3 audio alaw         alaw             (G.711 a-law)
   1 audio g723         g723             (G.723.1)
  19 audio speex        speex            (SpeeX)
  20 audio speex        speex16          (SpeeX 16khz)
  21 audio speex        speex32          (SpeeX 32khz)
  23 audio g722         g722             (G722)
  31 video h261         h261             (H.261 video)
  32 video h263         h263             (H.263 video)
   7 audio adpcm        adpcm            (Dialogic ADPCM)
  24 audio siren7       siren7           (ITU G.722.1 (Siren7, licensed from Polycom))
  40 audio silk         silk8            (SILK Codec (8 KHz))
  41 audio silk         silk12           (SILK Codec (12 KHz))
  42 audio silk         silk16           (SILK Codec (16 KHz))
  43 audio silk         silk24           (SILK Codec (24 KHz))
  27 audio g719         g719             (ITU G.719)
  33 video h263p        h263p            (H.263+ video)
  34 video h264         h264             (H.264 video)
  18 audio g729         g729             (G.729A)
   8 audio slin         slin             (16 bit Signed Linear PCM)
   9 audio slin         slin12           (16 bit Signed Linear PCM (12kHz))
  10 audio slin         slin16           (16 bit Signed Linear PCM (16kHz))
  11 audio slin         slin24           (16 bit Signed Linear PCM (24kHz))
  12 audio slin         slin32           (16 bit Signed Linear PCM (32kHz))
  13 audio slin         slin44           (16 bit Signed Linear PCM (44kHz))
  14 audio slin         slin48           (16 bit Signed Linear PCM (48kHz))
  15 audio slin         slin96           (16 bit Signed Linear PCM (96kHz))
  16 audio slin         slin192          (16 bit Signed Linear PCM (192kHz))
   2 audio ulaw         ulaw             (G.711 u-law)
  17 audio lpc10        lpc10            (LPC10)
  26 audio testlaw      testlaw          (G.711 test-law)
  39 audio none         none             (<Null> codec)
  25 audio siren14      siren14          (ITU G.722.1 Annex C, (Siren14, licensed from Polycom))
   6 audio g726aal2     g726aal2         (G.726 AAL2)
  36 video vp8          vp8              (VP8 video)
   4 audio gsm          gsm              (GSM)
  35 video mpeg4        mpeg4            (MPEG4 video)
  22 audio ilbc         ilbc             (iLBC)
  37 text  red          red              (T.140 Realtime Text with redundancy)
  38 text  t140         t140             (Passthrough T.140 Realtime Text)
  28 audio opus         opus             (Opus Codec)
  29 image jpeg         jpeg             (JPEG image)

Sadly, Opus is only going to be available on SNG7 and Asterisk 14 in the immediate future.

Luckily, the new SNG7 Iso is JUST ABOUT to be released.

I suggest you reconsider supporting Opus on SNG7/Asterisk 13. Many people would want to take advantage of Opus, but prefer to be on a long term support release of Asterisk.
Asterisk 14 is so fresh and only good for about a year, so we would prefer to stay on 13 and upgrade to 15 when it’s out.

Thank you.

It’s not available on Asterisk 13.

Looks like it is available on Asterisk 13. Quote from the article:
“Opus for Asterisk, it’s available now in Asterisk 14.0.1, and we’ll have a version of it available for Asterisk 13 in a couple of weeks, so that everyone using the current LTS release can take advantage, too.”

Release Notes of Asterisk 13.12.0:


[ASTERISK-25980] - [patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
[ASTERISK-26289] - Announcer channels in ConfBridges cause inefficiencies
[ASTERISK-26409] - codec_opus: Update Asterisk to support the translation codec.

We build asterisk with ‘enable everything’. As soon as Asterisk 13 starts producing the opus codec, it will appear in the RPM.

That is the results of ‘core show translation’ on asterisk 13. This is what happens on SNG7.

Talked to Digium on their forum as well. They are wondering why the codec is not there in 13.12.1, cause it should be, and are saying:
“The codec does not appear to be there. It’s definitely available as an option in “make menuselect” in 13.12.1 so the RPMs may not be doing what is needed to enable them or have the required dependency to do it via that method. I’ve made Rob aware.”

So looks like it should be possible to integrate Opus into Asterisk 13.

Ok well do what we always require and open a feature request and it can get in queue to look at adding. Priority right now is getting FreePBX 14 out to beta. It’s all hands on FreePBX 14 right now.

Just some clarification but we have decided internally to not support the Opus codec on FreePBX Distro 10.x

It will be supported in Asterisk 13 & 14 on Sangoma Distro 7.

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Confirmed working in SNG7 builds of Asterisk 13 as well as 14.

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